Re: [music-dsp] Fwd: Re: [dumb question] do Eurorack audio and CV signals use the same connectors?

[Laurent de Soras]

Laurie wrote:

Diode drops are not acceptable and a reversed diode needs to be rated to 
several amps at least.


You can use MOSFETs to avoid diode drops, although they are a bit
more expensive than diodes.




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Re: [music-dsp] Low cost DSPs

[Laurent de Soras]

Philippe Wicker wrote:


the A15 is not good at providing a sustained high cpu usage
without getting rapidly too hot.You rapidly get a throttling
cpu (the OS reduces the core frequency to a bare minimum
[...]
the code compiled in 64bit for the 1.5GHz C2 runs nearly as
fast than the same code compiled  in 32bit modes for the
1.8GHz RK3288, and 1.5 faster than the  32bit 1.2GHz PI3.


Good to know. I was wondering if I could replace my Pi 3 with
a XU4 but the heat issue is a no go and it seems that the C2 is
actually a better solution.


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Re: [music-dsp] Low cost DSPs

[Laurent de Soras]

Pablo Riera wrote:


I try some stuff on Rpi with supercollider + jack but I couldn't
manage to have lots of performance. As a synth it would crash
with more than 6 polyphony with 256 sample buffers. I think
the supercollider build  was messed up but need more
testing. What audio framework are you using?


I first tested Jack then directly attacked ALSA, but performance
was similar. My framework is just plain C++, with specific NEON
optimisations when possible. I don’t know about Supercollider
performance.

Also don’t forget that the the newer RPi 3 is much faster than
the first gen RPi for just a few more bucks. And if you need even
more horsepower the Odroid-XU4 looks pretty nice (and twice more
expensive).

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Re: [music-dsp] Low cost DSPs

[Laurent de Soras]

Does anyone has experience with some of these combinations
or others


Teensy + audio board is popular too.


and could share comments on technical issues, ease
of use, latency, audio quality (noise, max sampling rate), etc.


I’m working on something using a Raspberry Pi 3 + USB audio.
Latency is decent but not mindblowing : 13 ms for an audio
roundtrip with 64-sample buffers. You might have better
performances with I2S boards.

Audio quality depends on the audio interface. If the card is
powered by the USB port, watch for the noise coming from the
RPi.

For the ease of use, it’s a complete Linux system.

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Re: [music-dsp] ± 45° Hilbert transformer using pair of IIR APFs

[Laurent de Soras]

robert bristow-johnson wrote:


cool, so it appears from your comments to be real poles.
complex conjugate poles are not considered in the design.  i
wonder why that is?


I’m not the author of the formula, I just implemented it
and don’t understand how it produces these filters.
I guess the poles are real in both all-pass filter chains,
but once they are summed, the resulting filter exhibits
complex conjugate poles.




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Re: [music-dsp] ± 45° Hilbert transformer using pair of IIR APFs

[Laurent de Soras]

robert bristow-johnson wrote:


so Olli, how do you get your coefficients?  (if i may ask?)


If I may reply, I wrote long time ago a C++ library implementing
this kind of polyphase halfband filter, including a coefficient
calculator involving elliptic stuff (way beyond my knowledge).
Check the PolyphaseIir2Designer class:


There is also an ARM/NEON implementation below, not published in
the main release yet:


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Re: [music-dsp] Can anyone figure out this simple, but apparently wrong, mixing technique?

[Laurent de Soras]

robert bristow-johnson wrote:

it's this Victor Toth article: 
http://www.vttoth.com/CMS/index.php/technical-notes/68
and it doesn't seem to make sense to me.


The article is 16 year old now and mention even much older
technologies. I think that compression/limiting (or ducking?)
+ dithering is what he was actually looking for.

Anyway if you’re short on ideas for unusual Christmas gifts, you
can purchase a unrestricted license of his mixer for just $1500.


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Re: [music-dsp] Android related audio group / mailing list?

[Laurent de Soras]

Nuno Santos wrote:


I’m struggling with performance questions on Android.
https://groups.google.com/forum/#!topic/andraudio/zuXnnmZyePk


Your results tell that your program computes 10^9 operations
in a few dozen microseconds. There is definitely something
wrong here because even the most advanced ARM processors are
not known to be multi-teraflops CPUs.

Actually the compiler might think it can keep only one loop,
or even none, given that the result of each loop will always
be the same, and this result isn’t even used somewhere else
in the program. You should introduce slight dependencies in
your test in order to defeat this kind of optimisation.

The difference between the compiler settings you tried are
most likely caused by variations in initialisation sequences,
scheduling glitches or timer inaccuracies.

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Re: [music-dsp] Release pyo 0.8.0 (Python dsp library)

[Laurent de Soras]

list_em...@icloud.com wrote:

(Too bad the gifs
on the devmaster.net page have gone missing.)


I previously saved this page with all the graphics:


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Re: [music-dsp] Changing Biquad filter coefficients on-the-fly, how to handle filter state?

[Laurent de Soras]

Paul Stoffregen wrote:


Does anyone have any suggestions or publications or references to best
practices for what to do with the state variables of a biquad filter
when changing the coefficients?


Use an implementation designed to handle nicely coefficient
changes. For example the trapezoidal SVF:



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Re: [music-dsp] MSVC 2012/2013 upgrade with audio differences

[Laurent de Soras]

Stefan Sullivan wrote:


similar problems: the output is not bit-exact.


Have you a simple project showing the difference?

I’m not sure why you need this, but there are a lot of cases
where bit-exactness is not guaranteed when using floating
point data, especially if you compare FPU- and SSE2-generated
codes. Make sure that precision, rounding and behaviour with
denormals are set exactly the same in all the code paths.
Also, check the strictness of the Floating Point Model in
the code generation options.

--
Laurent de Soras  |   Ohm Force
DSP developer  Software designer |  Digital Audio Software
http://ldesoras.free.fr   | http://www.ohmforce.com

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Re: [music-dsp] IIR Coefficient Switching Glitches

[Laurent de Soras]

Chris Townsend wrote:


Any ideas?  Recommendations?


Probably this:
http://cytomic.com/files/dsp/SvfLinearTrapOptimised.pdf




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